Apparatus and method for encoding a signal as well as apparatus and method for decoding a signal

ABSTRACT

An apparatus and a method for encoding an input signal on the time base through orthogonal transform, comprising a step of removing the correlation of signal waveform on the basis of the parameters obtained by means of linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. The time base input signal from input terminal  10  is sent to normalization circuit section  11  and (LPC) analysis circuit  39 . The normalization circuit section  11  removes the correlation of the signal waveform and takes out the residue by means of LPC inverse filter  12  and pitch inverse filter  13  and sends the residue to orthogonal transform circuit section  25 . The LPC parameters from the lop analysis circuit  39  and the pitch parameters from the pitch analysis circuit  15  are sent to bit allocation calculation circuit  41 . Coefficient quantization section  40  quantizes the coefficients from the orthogonal transform circuit section  25  according to the number of allocated bits from the bit allocation calculation section  41.

BACKGROUND OF THE INVENTION

[0001] 1. Field of the Invention

[0002] This invention relates to an apparatus and a method for encodinga signal by quantizing an input signal through time base/frequency baseconversion as well as to an apparatus and a method for decoding anencoded signal. More particularly, the present invention relates to anapparatus and a method for encoding a signal that can be suitably usedfor encoding audio signals in a highly efficient way. It also relates toan apparatus and a method for decoding an encoded signal.

[0003] 2. Prior Art

[0004] Various methods for encoding an audio signal are known to dateincluding those adapted to compress the signal by utilizing statisticcharacteristics of audio signals (including voice signals and musicsignals) in terms of time and frequency and characteristic traits of thehuman hearing sense. Such coding methods can be roughly classified intoencoding in the time region, encoding in the frequency region andanalytic/synthetic encoding.

[0005] In the operation of transform coding of encoding an input signalon the time base by orthogonally transforming it into a signal on thefrequency base, it is desirable from the viewpoint of coding efficiencythat the characteristics of the time base waveform of the input signalare removed before subjecting it to transform coding.

[0006] Additionally, when quantizing the coefficient data on theorthogonally transformed frequency base, the data are more often thannot weighted for bit allocation. However, it is not desirable totransmit the information on the bit allocation as additional informationor side information because it inevitably increases the bit rate.

[0007] In view of these circumstances, it is therefore an object of thepresent invention to provide an apparatus and a method for encoding asignal that are adapted to remove the characteristic or correlativeaspects of the time base waveform prior to orthogonal transform in orderto improve the coding efficiency and, at the same time, reduce the bitrate by making the corresponding decoder able to know the bit allocationwithout directly transmitting the information on the bit allocation usedfor the quantizing operation.

[0008] Meanwhile, for the operation of transform coding of encoding aninput signal on the time base by orthogonally transforming it into asignal on the frequency base, techniques have been proposed to quantizethe coefficient data on the frequency base by dynamically allocatingbits in response to the input signal in order to realize a low codingrate. However, cumbersome arithmetic operations are required for the bitallocation particularly when the bit allocation changes for eachcoefficient in the operation of dividing coefficient data on thefrequency base in order to produce sub-vectors for vector quantization.

[0009] Additionally, the reproduced sound can become highly unstablewhen the bit allocation changes extremely for each frame that provides aunit for orthogonal transform.

[0010] In view of these circumstances, it is therefore another object ofthe present invention to provide an apparatus and a method for encodinga signal that are adapted to dynamically allocate bits in response tothe input signal with simple arithmetic operations for the bitallocation and reproduce sound without making it unstable if the bitallocation changes remarkably among frames for the operation of encodingthe input signal that involves orthogonal transform as well as anapparatus and a method for decoding a signal encoded by such anapparatus and a method.

[0011] Additionally, since quantization takes place after the bitallocation for the coefficient on the frequency base such as the MDCTcoefficient in the operation of transform coding of encoding an inputsignal on the time base by orthogonally transforming it into a signal onthe frequency base, quantization errors spreads over the entireorthogonal transform block length on the time base to give rise to harshnoises such as pre-echo and post-echo. This tendency is particularlyremarkable for sounds that relatively quickly attenuate between pitchpeaks. This problem is conventionally addressed by switching thetransform window size (so-called window switching). However, thistechnique of switching the transform window size involves cumbersomeprocessing operations because it is not easy to detect the right windowhaving the right size.

[0012] In view of the above circumstances, it is therefore still anotherobject of the present invention to provide an apparatus and a method forencoding a signal adapted to reduce harsh noises such as pre-echo andpost-echo without modifying the transform window size as well as anapparatus and a method for decoding a signal encoded by such anapparatus and a method.

SUMMARY OF THE INVENTION

[0013] According to a first aspect of the invention, the above objectsare achieved by providing a method for encoding an input signal on thetime base through orthogonal transform, said method comprising:

[0014] a step of removing the correlation of signal waveform on thebasis of the parameters obtained by means of linear predictive coding(LPC) analysis and pitch analysis of the input signal on the time baseprior to the orthogonal transform.

[0015] Preferably, the input time base signal is transformed tocoefficient data on the frequency base by means of modified discretecosine transform (MDCT) in said orthogonal transform step. Preferably,in said normalization step, the LPC analysis residue of said inputsignal is output on the basis of the LPC coefficient obtained throughLPC analysis of said input signal and the correlation of the pitch ofsaid LPC prediction residue is removed on the basis of the parametersobtained through pitch analysis of said LPC prediction residue.Preferably, said quantization means quantizes according to the number ofallocated bits determined on the basis of the outcome of said LPCanalysis and said pitch analysis.

[0016] According to a second aspect of the invention, there is provideda method for encoding an input signal on the time base throughorthogonal transform, said method comprising:

[0017] a calculating step of calculating weights as a function of saidinput signal; and

[0018] a quantizing step of determining an order for the coefficientdata obtained through the orthogonal transform according to the order ofthe calculated weights and carrying out an accurate quantizing operationaccording to the determined order.

[0019] Preferably, in said quantizing step, a larger number of allocatedbits are used for quantization for the coefficient data of a higherorder.

[0020] Preferably, the coefficient data obtained through said orthogonaltransform are divided into a plurality of bands on the frequency baseand the coefficient data of each of the bands are quantized according tosaid determined order of said weights independently from the remainingbands.

[0021] Preferably, the coefficient data of each of the bands are dividedinto a plurality of groups in the descending order of the bands todefine respective coefficient vectors and each of the obtainedcoefficient vectors is subjected to vector quantization.

[0022] According to a third aspect of the invention, there is provided amethod for encoding an input signal on the time base through orthogonaltransform on a frame by frame basis, each frame providing a coding unit,said method comprising:

[0023] an envelope extracting step of an extracting envelope within eachframe of said input signal; and

[0024] a gain smoothing step of carrying out a gain smoothing operationon said input signal on the basis of the envelope extracted by saidenvelope extracting step and supplying the input signal for saidorthogonal transform.

[0025] Preferably, the input time base signal is transformed tocoefficient data on the frequency base by means of modified discretecosine transform (MDCT) for said orthogonal transform. Preferably, theinformation on said envelope is quantized and output. Preferably, saidframe is divided into a plurality of sub-frames and said envelope isdetermined as the root means square (rms) value of each of the dividedsub-frames. Preferably, the rms value of each of the divided sub-framesis quantized and output.

[0026] Thus, according to the first aspect of the invention, there isprovided a method for encoding an input signal on the time base throughorthogonal transform, said method comprising:

[0027] a step of removing the correlation of signal waveform on thebasis of the parameters obtained by means of linear predictive coding(LPC) analysis and pitch analysis of the input signal on the time baseprior to the orthogonal transform.

[0028] With this arrangement, a residual signal that resembles a whitenose is subjected to orthogonal transform to improve the codingefficiency. Additionally, in a method for encoding an input signal onthe time base through orthogonal transform, preferably a quantizationoperation is conducted according to the number of allocated bitsdetermined on the basis of the outcome of said linear predictive coding(LPC) analysis and said pitch analysis. Then, the corresponding decoderis able to reproduce the bit allocation of the encoder from theparameters of the LPC analysis and the pitch analysis to make itpossible to suppress the rate of transmitting side information and hencethe overall bit rate and improve the coding efficiency.

[0029] Still additionally, the operation of encoding high quality audiosignals can be carried out highly efficiently by using a technique ofmodified discrete cosine transform (MDCT) for orthogonal transform.

[0030] According to the second aspect of the invention, there isprovided a method for encoding an input signal on the time base throughorthogonal transform, said method comprising:

[0031] a calculating step of calculating weights as a function of saidinput signal; and

[0032] a quantizing step of determining an order for the coefficientdata obtained through the orthogonal transform according to the order ofthe calculated weights and carrying out an accurate quantizing operationaccording to the determined order.

[0033] With this arrangement, it is possible to dynamically allocatebits in response to the input signal with simple arithmetic operationsfor calculating the number of bits to be allocated to each coefficient.

[0034] Particularly, when the coefficient data obtained through saidorthogonal transform are divided into a plurality of sub-vectors, thenumber of bits to be allocated to each sub-vector can be determined bycalculating the weight for it to reduce the arithmetic operations if thenumber of bits to be allocated to each coefficient changes because thecoefficient data can be reduced into sub-vectors after they are sortedout according to the descending order of the weights.

[0035] Additionally, when the coefficient data on the frequency base aredivided into bands and the number of bits to be allocated to each bandis predetermined, any possible abrupt change in the quantizationdistortion can be prevented from taking place to reproduce sound on astable basis if the weight of each coefficient change extremely fromframe to frame because the number of allocated bits is reliabledetermined for each band.

[0036] Still additionally, when the parameters to be used for thearithmetic operations of bit allocation are predetermined andtransmitted to the decoder, it is no longer necessary to transmit theinformation on bit allocation to the decoder so that it is possible tosuppress the rate of transmitting side information and hence the overallbit rate and improve the coding efficiency. Still additionally, theoperation of encoding high quality audio signals can be carried outhighly efficiently by using a technique of modified discrete cosinetransform (MDCT) for orthogonal transform.

[0037] According to the third aspect of the invention, there is provideda method for encoding an input signal on the time base throughorthogonal transform on a frame by frame basis, each frame providing acoding unit, said method comprising:

[0038] an envelope extracting step of an extracting envelope within eachframe of said input signal; and

[0039] a gain smoothing step of carrying out a gain smoothing operationon said input signal on the basis of the envelope extracted by saidenvelope extracting step and supplying the input signal for saidorthogonal transform.

[0040] With this arrangement, it is possible to reduce harsh noises suchas pre-echo and post-echo without modifying the transform window size asin the case of the prior art.

[0041] Additionally, when the information on said envelope is quantizedand output to the decoder and the gain is smoothed by using thequantized envelope value, the decoder can accurately restore the gain.

[0042] Still additionally, the operation of encoding high quality audiosignals can be carried out highly efficiently by using a technique ofmodified discrete cosine transform (MDCT) for orthogonal transform.

BRIEF DESCRIPTION OF THE DRAWINGS

[0043]FIG. 1A is a schematic block diagram of an embodiment of encoderaccording to the first aspect of the invention.

[0044]FIG. 1B is a schematic block diagram of a quantization circuitthat can be used for an embodiment of encoder according to the secondaspect of the invention.

[0045]FIG. 1C is a schematic block diagram of an embodiment of encoderaccording to the third aspect of the invention.

[0046]FIG. 2 is a schematic block diagram of an audio signal encoder,which is a specific embodiment of the invention.

[0047]FIG. 3 is a schematic illustration of the relationship between aninput signal and an LPC analysis and a pitch analysis conducted for it.

[0048]FIGS. 4A through 4C are schematic illustrations of a time basesignal waveform for illustrating how the correlation of signal waveformis removed by an LPC analysis and a pitch analysis conducted on a timebase input signal.

[0049]FIGS. 5A through 5C are schematic illustrations of frequencycharacteristics illustrating how the correlation of signal waveform isremoved by an LPC analysis and a pitch analysis conducted on a time baseinput signal.

[0050]FIG. 6 is a schematic illustration of a time base input signalillustrating an overlap-addition of a decoder.

[0051]FIGS. 7A through 7C are schematic illustrations of a sortingoperation based on the weights of coefficients within a band obtained bydividing coefficient data.

[0052]FIG. 8 is a schematic illustration of an operation ofvector-quantization of dividing each coefficient sorted out according tothe weight within a band obtained by dividing coefficient data intosub-vectors.

[0053]FIG. 9 is a schematic block diagram of an embodiment of audiosignal decoder corresponding to the audio signal encoder of FIG. 2.

[0054]FIG. 10 is a schematic block diagram of an inverse quantizationcircuit that can be used for the audio signal decoder of FIG. 9.

[0055]FIG. 11 is a schematic block diagram of an embodiment of decodercorresponding to the encoder of FIG. 1C.

[0056]FIG. 12 is a schematic illustration of a reproduced signalwaveform that can be obtained by encoding a sound of a castanet withoutgain control.

[0057]FIG. 13 is a schematic illustration of a reproduced signalwaveform that can be obtained by encoding a sound of a castanet withgain control.

[0058]FIG. 14 is a schematic illustration of the waveform of a time basesignal in an initial stage of the speech burst of part of a soundsignal.

[0059]FIG. 15 is a schematic illustration of the frequency spectrum inan initial stage of the speech burst of part of a sound signal.

DETAILED DESCRIPTION OF THE INVENTION

[0060] Now, the present invention will be described in greater detail byreferring to the accompanying drawings that illustrate preferredembodiments of the invention.

[0061]FIG. 1A is a schematic block diagram of an embodiment of encoderaccording to the first aspect of the invention.

[0062] Referring to FIG. 1A, a waveform signal on the time base such asa digital audio signal is applied to input terminal 10. While a specificexample of such a digital audio signal may be a so-called broad bandsound signal with a frequency band between 0 and 8 kHz and a samplingfrequency Fs of 16 kHz, although the present invention is by no meanslimited thereto.

[0063] The input signal is then sent from the input terminal 10 tonormalization circuit section 11. The normalization circuit section 11is also referred to as whitening circuit and adapted to carry out awhitening operation of extracting characteristic traits of the inputtemporal waveform signal and taking out the prediction residue. Atemporal waveform can be whitened by way of linear or non-linearprediction. For example, an input temporal waveform signal can bewhitened by way of LPC (linear predictive coding) analysis and pitchanalysis.

[0064] Referring to FIG. 1, the normalization (whitening) circuitsection 11 comprises an LPC inverse filter 12 and a pitch inverse filter13. The input signal entered through the input terminal 10 is sent tothe LPC analysis circuit 39 for LPC analysis and the LPC coefficients(so-called α parameters) obtained as a result of the analysis are sentto the pitch inverse filter 13 in order to take out the processingresidue. The LPC prediction residue from the LPC inverse filter 12 isthen sent to pitch analysis circuit 15 and the pitch inverse filter 13.The pitch parameters are taken out by the pitch analysis circuit 15 byway of pitch analysis, which will be described hereinafter, and thepitch correlation is removed by the pitch inverse filter 13 from saidLPC predictive residue to obtain the pitch residue, which is then sentto the orthogonal transform circuit 25. The LPC coefficients from theLPC analysis circuit 39 and the pitch parameters from the pitch analysiscircuit 15 are then sent to bit allocation calculating circuit 41, whichis adapted to determine the bit allocation for the purpose ofquantization.

[0065] The whitened temporal waveform signal, which is the pitch residueof the LPC rotary speed, sent from the normalization circuit section 11is by turn sent to orthogonal transform circuit section 25 for timebase/frequency base transform (T/F mapping), where it is transformedinto a signal (coefficient data) on the frequency base). Techniques thatare popularly used for the T/F mapping include DCT (discrete cosinetransform), MDCT (modified discrete cosine transform) and FFT (fastFourier transform). The parameters, or the coefficient data, such as theMDCT coefficients or the FFT coefficients obtained from the orthogonaltransform circuit section 25 are then sent to the coefficient quantizingsection 40 for SQ (scalar quantization) or VQ (vector quantization). Itis necessary to determine a bit allocation for each coefficient for thepurpose of quantization if the operation of coefficient quantization isto be carried out efficiently. The bit allocation can be determined onthe basis of a hearing sense masking model, various parameters such asthe LPC coefficients and pitch parameters obtained as a result of thewhitening operation of the normalization circuit section 11 or the Barkscale factors calculated from the coefficient data. The Bark scalefactor typically include the peak values or the rms (root mean square)values of each critical band obtained when the coefficients determinedas a result of the orthogonal transform are divided to critical bands,which are frequency bands wherein a greater band width is used for ahigher frequency band to correspond to the characteristic traits of thehuman hearing sense.

[0066] In this embodiment, the bit allocation is defined in such a waythat it is determined only on the basis of LPC coefficients, pitchparameters and Bark scale factors so that the decoder can reproduce thebit allocation of the encoder when the former receives only theseparameters. Then, it is no longer necessary to transmit additionalinformation (side information) including the number of allocated bitsand hence the transmission bit rate can be reduced significantly.

[0067] Note that quantized values are used for the LPC coefficients (αparameters) to be used in the LPC inverse filter and the (pitch gainsof) the pitch parameters to be used in the pitch inverse filter 13 fromthe viewpoint of the reproducibility of the decoder.

[0068]FIG. 1B is a schematic block diagram of a quantization circuitthat can be used for an embodiment of encoder according to the secondaspect of the invention.

[0069] Referring to FIG. 1B, input terminal 1 is fed with thecoefficient data on the frequency base obtained by orthogonallytransforming a time base signal and weight calculation circuit 2 is fedwith parameters such as LPC coefficients, pitch parameters and Barkscale factors. The weight calculation circuit 2 calculates weights w onthe basis of such parameters. In the following description, thecoefficients of a frame of orthogonal transform is expressed by vector yand the weights of a frame of orthogonal transform is expressed byvector w.

[0070] The coefficient vector y and the weight vector w are then sent toband division circuit 3, which divides them among L (L≧1) bands. Thenumber of bands may typically be three (L=3) including a low band, amiddle band and a high band, although the present invention is by nomeans limited thereto. It is also possible not to divide them amongbands for the purpose of the invention. If the coefficient vector andthe weight vector of the k-th band are y_(k) and w_(k) respectively(0≦k≦L−1), the following formulas are obtained.

y=(y ₀ , y ₁ , . . . , y _(L−1))

w=(w ₀ , w ₁ , . . . , w _(L−1))

[0071] The number of bands used for dividing the coefficients and theweights and the number of coefficients of each band are set topredetermined respective values.

[0072] Then, the coefficient vectors=y₀, y₁, . . . , y_(L−1) are sent torespective sorting circuits 4 ₀, 4 ₁, . . . , 4 _(L−1) and thecoefficients in each band is provided with respective order numbers inthe descending order of the weights. This operation may be carried outeither by rearranging (sorting) the coefficients themselves in the bandin the descending order of the weights or by sorting the indexes of thecoefficients indicating their respective positions on the frequency basein the descending order of the weights and determining the accuracylevel (the number of allocated bits) of each coefficient to reflect thesorted index of the coefficient at the time of quantization. Whenrearranging the coefficients themselves, the coefficient vector y′_(k)whose coefficient s are sorted in the descending order of the weightscan be obtained by sorting the coefficients of the coefficient vectory_(k) of the k-th band in the descending order of the weights.

[0073] Then, the coefficient vectors y₀, y₁, . . . , y_(L−1), thecoefficients of each of which are sorted in the descending order of theweights of the band, are then sent to respective vector quantizers 5 ₀,5 ₁, . . . , 5 _(L−1), where they are subjected to respective operationsof vector-quantization.

[0074] Then, the vectors c₀, c₁, . . . , c_(L−1) of the coefficientindexes of the bands sent from the respective vector quantizers 5 ₀, 5₁, , , , , 5 _(L−1) are collectively taken out as vector c of thecoefficient indexes of all the bands.

[0075] The operation of the quantization circuit of FIG. 1B will bedescribed in greater detail by referring to FIGS. 7 and 8.

[0076] With the above arrangement, the coefficients that are sorted inthe descending order of the weights can be sequentially subjected torespective operations of vector-quantization if the weights of thecoefficients of each frame change dynamically so that the process of bitallocation can be significantly simplified. Additionally, if the numberof bits allocated to each band is fixed and hence invariable., thensound can be reproduced on a stable basis even if weights changessignificantly among frames for the signal.

[0077]FIG. 1C is a schematic block diagram of an embodiment of encoderaccording to the third aspect of the invention.

[0078] Referring to FIG. 1C, a waveform signal on the time base, whichis typically a digital audio signal, is entered to input terminal 9.While a specific example of such a digital audio signal may be aso-called broad band sound signal with a frequency band between 0 and 8kHz and a sampling frequency Fs of 16 kHz, although the presentinvention is by no means limited thereto. The prediction residueobtained by extracting characteristic traits of a temporal waveformsignal by means of a normalization circuit (whitening circuit) may beused for the time base input signal.

[0079] The signal from the input terminal 9 is then sent to envelopeextraction circuit 17 and windowing circuit 26. The envelope extractioncircuit 17 extracts envelopes within each frame that operates as acoding unit of MDCT (modified discrete cosine transform) circuit 27,which is an orthogonal transform circuit. More specifically, it dividesa frame into a plurality of sub-frames and calculates the root meansquare (rms) for each sub-frame as envelope. The obtained envelopeinformation is quantized by the quantizer 20 and the obtained index(envelope index) is taken out from output terminal 21 and sent to thedecoder.

[0080] In the windowing circuit 26, an window-placing operation iscarried out by means of a window function that can utilize aliasingcancellation of MDCT through ½ overlapping. The output of the windowingcircuit 26 is divided by divider 14 that operates as gain smoothingmeans, using the value of the envelope quantized by the quantizer 20 asdivisor. Then, the obtained quotient is sent to the MDCT circuit 27. Thequotient is transformed into coefficient data (MDCT coefficients) on thefrequency base by the MDCT circuit 27 and the obtained MDCT coefficientsare quantized by quantization circuit section 40 and the indexes of thequantized MDCT coefficients are then taken out from output terminal 51and sent to the decoder. Note that the orthogonal transform is notlimited to MDCT for the purpose of the invention.

[0081] With the above arrangement, a noise shaping process proceedsalong the time base so that quantized noises that is harsh to the earsuch as pre-echo can be reduced without switching the transform widowsize.

[0082] While the embodiments of signal encoder of FIGS. 1A, 1B and 1Care illustrated as hardware, they may alternatively be realized assoftware by means of a so-called DSP (digital signal processor).

[0083] Now, the present invention will be described in greater detail byway of a specific example illustrated in FIG. 2, which is an audiosignal encoder.

[0084] The audio signal encoder of FIG. 2 is adapted to carry out anoperation of time base/frequency base transform (T/F transform), whichmay be MDCT (modified discrete cosine transform), on the supplied timebase signal by means of the orthogonal transform section 2. In theillustrated example, characteristic traits of the input signal waveformof the time base signal are extracted by way of LPC analysis, pitchanalysis and envelope extraction before the orthogonal transform and theparameters expressing the extracted characteristic traits areindependently quantized and taken out. Then, the parameters expressingthe characteristic traits are quantized separately and taken out.Subsequently, the characteristic traits and the correlation of thesignal are removed by the normalization (whitening) circuit section 11to produce a noise-like signal that resembles white noise in order toimprove the coding efficiency.

[0085] The LPC coefficients obtained by the above LPC analysis and thepitch parameters obtained by the above pitch analysis are used fordetermining the bit allocation for the purpose of quantization ofcoefficient data after the orthogonal transform. Additionally, Barkscale factors obtained as normalization factors by taking out the peakvalues and the rms values of the critical bands on the frequency basemay also be used. In this way, the weights to be used for quantizing theorthogonal transform coefficient data such as MDCT coefficients arecomputationally determined by means of the LPC coefficients, the pitchparameters and the Bark scale factors and then bit allocation isdetermined for all the bands to quantize the coefficients. When theweights to be used for quantization are determined by preselectedparameters such as LPC coefficients, pitch parameters and Bark scalefactors as described above, the decoder can exactly reproduce the bitallocation of the encoder simply by receiving the parameters so that itis no longer necessary to transmit the side information on the bitallocation per se.

[0086] Additionally, when quantizing coefficients, the coefficient dataare rearranged (sorted) in the order of the weights or the allocatednumbers of bits to be used for the quantizing operation in order tosequentially and accurately quantize the coefficient data. Thisquantizing operation is preferably carried out by dividing the sortedcoefficients sequentially from the top into sub-vectors so that thesub-vectors may be quantizes independently. While the coefficient dataof the entire band may be sorted, they may alternatively be divided intoa number of bands so that the sorting operation may be carried out on aband by band basis. Then, only if the parameters to be used for the bitallocation are preselected, the decoder can exactly reproduce the bitallocation and the sorting order of the encoder by receiving theparameters and not receiving the information on the bit allocation andthe positions of the sorted coefficients.

[0087] Referring to FIG. 2, a digital audio signal obtained by A/Dtransforming a broad band audio input signal with a frequency bandtypically between 0 and 8 kHz, using a sampling frequency Fs=16 kHz, isapplied to the input terminal 10. The input signal is sent to LPCinverse filter 12 of normalization (whitening) circuit section 11 and,at the same time, taken by every 1024 samples, for example, and sent toLPC analysis/quantization section 30. The LPC analysis/quantizationsection 30 carries out a hamming/window-placing operation on the inputsignal and computationally determines LPC coefficients of the 20th orderor so, which are α parameters, so that the LPC residue may be obtainedby the LPC inverse filter 11. During this operation of LPC analysis,part of the 1024 samples of a frame that provide a unit of analysis,e.g., a half of them or 512 samples, are made to overlap the next blockto make the frame interval equal to 512 samples. This arrangement isused to utilize the aliasing cancellation of the MDCT employed for thesubsequent orthogonal transform. The LPC analysis/quantization section30 is adapted to transmit the α parameters, which are LPC coefficients,after transforming them into LSP (linear spectral pair) parameters andquantizing them.

[0088] The α parameters from LPC analysis circuit 32 are sent to α→LSPtransform circuit 33 and transformed into linear spectral pair (LSP)parameters. This circuit transforms the α parameters obtained as directtype filter coefficients into 20, or 10 pairs of, LSP parameters. Thistransforming operation is carried out typically by means of theNewton-Rapson method. This operation of transforming α parameters intoLSP parameters is carried out because the latter are more excellent thanthe former in terms of interpolation effect.

[0089] The LSP parameters from the α→LSP transform circuit 33 arevector-quantized or matrix-quantized by LSP quantizer 34. At this time,they may be subjected to vector-quantization after determining theinter-frame differences or the LSP parameters of a plurality of framesmay be collectively matrix-quantized.

[0090] The quantized output of the LSP quantizer 34 are the indexes ofthe LSP vector-quantization and taken out by way of terminal 31, whereasthe quantized LSP vectors or the inverse quantization outputs are sentto LSP interpolation circuit 36 and LSP→α transform circuit 38.

[0091] The LSP interpolation circuit 36 interpolates the immediatelypreceding frame and the current frame of the LSP vector quantized by theLSP quantizer 34 on a frame by frame basis to obtain the rate requiredin subsequent processing steps. In this embodiment, it operates forinterpolation at a rate 8 times as high as the original rate.

[0092] Then, the LSP→α transform circuit 37 transforms the LSPparameters into α parameters that are typically coefficients of the 20thorder of a direct type filer in order to carry out an inverse filteringoperation of the input sound by means of the interpolated LSP vector.The output of the LSP→α transform circuit 37 is then sent to LPC inversefilter circuit 12 adapted to determine the LPC residue. The LPC inversefilter circuit 12 carries out an inverse filtering operation by means ofthe α parameters that are updated at a rate 8 times as high as theoriginal rate in order to produce a smooth output.

[0093] On the other hand, the LSP coefficients that are sent from theLSP quantization circuit 34 and updated at the original rate are sent toLSP→α transform circuit 38 and transformed into α parameters, which arethen sent to bit allocation determining circuit 41 for determining thebit allocation. The bit allocation determining circuit 41 alsocalculates the weights w(ω) to be used to quantizing MDCT coefficientsas will be described hereinafter.

[0094] The output from the LPC inverse filter 12 of the normalization(whitening) circuit section 11 is then sent to the pitch inverse filter13 and the pitch analysis circuit 15 for pitch prediction, that is along term prediction.

[0095] Now, a long term prediction will be discussed below. A long termprediction is an operation of determining the pitch prediction residuewhich is the difference obtained by subtracting the waveform displacedon the time base by a pitch period or a pitch lag obtained as a resultof pitch analysis from the original waveform. In this example, atechnique of three-point prediction is used for the long termprediction. The pitch lag refers to the number of samples correspondingto the pitch period of the sampled time base data.

[0096] Thus, the pitch analysis circuit 15 carries out a pitch analysisonce for every frame to make the analysis cycle equal to a frame. Thepitch lag obtained as a result of the pitch analysis is sent to thepitch inverse filter 13 and the bit allocation determining circuit 41,while the obtained pitch gain is sent to pitch gain quantizer 16. Thepitch lag index obtained by the pitch analysis circuit 15 is taken outfrom terminal 52 and sent to the decoder.

[0097] The pitch gain quantizer 16 vector-quantizes the pitch gainsobtained at three points corresponding to the above three-pointprediction and the obtained code book index (pitch gain index) is takenout from output terminal 53. Then, the vector of the representativevalue or the inverse quantization output is sent to the pitch inversefilter 13. The pitch inverse filter 13 output the pitch predictionresidue of the three-point prediction on the basis of the abovedescribed pitch analysis. The pitch prediction residue is sent to thedivider 14 and the envelope extraction circuit 17.

[0098] Now, the pitch analysis will be described further. In the pitchanalysis, pitch parameters are extracted by means of the above LPCresidue. A pitch parameter comprises a pitch lag and a pitch gain.

[0099] Firstly, the pitch lag will be determined. For example, a totalof 512 samples are cut out from a central portion of the LPC residue andexpressed by x(n) (n=0˜511) or x. If the 512 samples of the k-th LPCresidue as counted back from the current LPC residue is expressed byx_(k), the pitch k is defined as a value that minimizes

∥x−gx_(1k)∥².

[0100] Thus, if

g=(x, x _(k))² /∥x _(k)∥²,

[0101] an optimal lag K can be obtained by searching for k thatmaximizes

(x, x_(k))²/∥x_(k)∥².

[0102] In this embodiment, 12≦K≦240. This K may be used directly or,alternatively, a value obtained by means of a tracking operation usingthe pitch lag of past frames may be used. Then, by using the obtained K,an optimal pitch gain will be determined for each of three points (K,K−1, K+1). In other words, g⁻¹, g₀ and g₁ that minimize

|x−(g⁻¹ x_(L+1)+g₀ x_(L)+g₁ x_(L−1))∥²

[0103] will be determined and selected as pitch gains for the threepoints. The pitch gains of the three points are sent to the pitch gainquantizer 16 and collectively vector-quantized. Then, quantized pitchgain and the optimal lag K are used for the pitch inverse filter 13 todetermine the pitch residue. The obtained pitch residue is linked to thepast pitch residues that are already known and then subjected to an MDCTtransform operation as will be discussed in greater detail hereinafter.The pitch residue may be held under time base gain control prior to theMDCT transform.

[0104]FIG. 3 is a schematic illustration of the relationship between aninput signal and an LPC analysis and a pitch analysis conducted for it.Referring to FIG. 3, the analysis cycle of a frame FR, from which 1,024samples may be taken, has a length corresponding to an MDCT transformblock. In FIG. 3, time t₁ indicates the center of the current and newLPC analysis (LSP₁) and time t₀ indicates the center of the LPC analysis(LSP₀) of the immediately preceding frame. The latter half of thecurrent frame contains new data ND, whereas the former half of thecurrent frame contains previous data PD. In FIG. 3, a denotes the LPCresidue obtained by interpolating LSP₀ and LSP₁ and b denotes the LPCresidue of the immediately preceding frame, while c denotes the newpitch residue obtained by the pitch analysis using this portion (latterhalf of b+former half of a) as object and d denotes the pitch residue ofthe past. Referring to FIG. 3, a can be determined at the time when allthe new data ND are input and the new pitch residue c can becomputationally determined from a and b that is already known. Then, thedata FR of the frame to be subjected to orthogonal transform areprepared by linking c and the pitch residue d that is already known. Thedata FR of the frame are then actually subjected to orthogonal transformthat may be MDCT transform.

[0105]FIGS. 4A through 4C are schematic illustrations of a time basesignal waveform for illustrating how the correlation of signal waveformis removed by an LPC analysis and a pitch analysis conducted on a timebase input signal. FIG. 5 are schematic illustrations of frequencycharacteristics illustrating how the correlation of signal waveform isremoved by an LPC analysis and a pitch analysis conducted on a time baseinput signal. More specifically, FIG. 4(A) shows the waveform of theinput signal and FIG. 5(A) shows the frequency spectrum of the inputsignal. Then, the characteristic traits of the waveform are extractedand removed by using an LPC inverse filter formed on the basis of theLPC analysis to produce a time base waveform (LPC residue waveform)showing the form of a substantially periodical pulse as shown in FIG.4(B). FIG. 5(B) shows the frequency spectrum corresponding to the LPCresidue waveform. Then, the pitch components are extracted and removedfrom the LPC residue by using a pitch inverse filter formed on the basisof the pitch analysis to produce a time base signal that resembles whitenoise (noise-like) as shown in FIG. 4(C). FIG. 5(C) shows the frequencyspectrum corresponding to the time base signal of FIG. 4(C).

[0106] In the above embodiment of the invention, the gains of the datawithin the frame are smoothed by means of the normalization (whitening)circuit section 11. This is an operation of extracting an envelope fromthe time base waveform in the frame (the residue of the pitch inversefilter 13 of this embodiment) by means of the envelope extractioncircuit 17, sending the extracted envelope to envelope quantizer 20 byway of switch 19 and dividing the time base waveform (the residue of thepitch inverse filter 13) by the value of the quantized envelope by meansof the divider 14 to produce a signal smoothed on the time base. Thesignal produced by the divider 14 is sent to the downstream orthogonaltransform circuit section 25 as output of the normalization (whitening)circuit section 11.

[0107] With this smoothing operation, it is possible to realize anoise-shaping of causing the size of the quantization error producedwhen inversely transforming the quantized orthogonal transformcoefficients into a temporal signal to follow the envelope of theoriginal signal.

[0108] Now, the operation of extracting an envelope of the envelopeextraction circuit 17 will be discussed below. If the signal supplied tothe envelope extraction circuit 17, which is the residue signalnormalized by the LPC inverse filter 12 and the pitch inverse filter 13,is expressed by x(n), n=0˜N−1 (N being the number of samples of a frameFR, or the orthogonal transform window size, e.g., N=1,024), the valueof rms (root mean square) of the sub-blocks or the sub-frames producedby dividing it by a length M shorter than the transform window size N,e.g., M=N/8, is used for the envelope. In other words, the value ofrms_(i) of the i-th sub-block (i=0˜M−1) that is normalized is defined byformula (1) below. $\begin{matrix}{{r\quad m\quad s_{i}} = \sqrt{\frac{\sum\limits_{k = 0}^{M - 1}{{x\left( {{iM} + k} \right)}{x\left( {{iM} + k} \right)}}}{\frac{M}{\frac{\sum\limits_{k = 0}^{N - 1}{{x(k)}{x(k)}}}{N}}}}} & (1)\end{matrix}$

[0109] Then, each of rms_(i) obtained from formula (a) can bescalar-quantized or rms_(i) can be collectively vector-quantized as asingle vector. In this embodiment, rms_(1i) is collectivelyvector-quantized and the index is taken out from terminal 21 asparameter to be used for the purpose of time base gain control or asenvelope index and transmitted to the decoder.

[0110] The quantized rms_(i) of each sub-block (sub-frame) is expressedby qrms_(i) and the input residue signal x(n) is divided by qrms_(i) bymeans of the divider 14 to obtain signal x_(g) (n) that is smoothed onthe time base. If, of the values of rms_(i) obtained in this way, theratio of the largest one to the smallest one is equal to or greater thana predetermined value (e.g., 4), they are subjected gain control asdescribed above and a predetermined number of bits (e.g., 7 bits) areallocated for the purpose of quantizing the parameters (the abovedescribed envelope indexes). However, if the ratio of the largest one tothe smallest one of the values of rms_(i) of each sub-block (sub-frame)of the frame is smaller than the predetermined value, they are allocatedfor the purpose of quantization of other parameters such as frequencybase parameters (orthogonal transform coefficient data). The judgment ifa gain control operation is carried out or not is made by gain controlon/off judgment circuit 18 and the result of the judgment (gain controlswitch SW) is transmitted as switching control signal to the input sideswitch 19 of the envelope quantization circuit 20 and also to thecoefficient quantization circuit 45 in the coefficient quantizationsection 40, which will be described in greater detail hereinafter, andused for switching from the number of bits allocated to the coefficientfor the on state of the gain control to the coefficient for the offstate of the gain control or vice versa. The result of the judgment(gain control switch SW) of the gain control on/off judgment circuit isalso taken out byway of terminal 22 and sent to the decoder.

[0111] The signals x_(s) (n) that are controlled (compressed) for thegain by the divider 14 and smoothed on the time base are then sent tothe orthogonal transform circuit section 25 as output of thenormalization circuit section 11 and transformed into frequency baseparameters (coefficient data) typically by means of MDCT. The orthogonaltransform circuit section 25 comprises a windowing circuit and an MDCTcircuit 27. In the windowing circuit 26, they are subjected to awindow-placing operation of a window function that can utilize aliasingcancellation of MDCT on the basis of ½frame overlap.

[0112] When decoding the signal at the side of the decoder, the decoderinversely quantizes the transmitted quantization indexes of thefrequency base parameters (e.g., MDCT coefficients). Subsequently, anoperation of overlap-addition and a operation (gain expansion or gainrestoration) that is inverse relative to the smoothing operation forencoding are conducted by using the inversely quantized time base gaincontrol parameters. It should be noted that the following process has tobe followed when the technique of gain smoothing is used because nooverlap-addition can be used by utilizing an virtual window, with whichthe square sum of the window value of an ordinarily symmetric andoverlapping position is held to a constant value.

[0113]FIG. 6 is a schematic illustration of a time base input signalillustrating an overlap-addition and gain control of a decoder.Referring to FIG. 6, w(n), n=0˜N−1 represents an analysis/synthesiswindow and g(n) represents time base gain control parameters. Thus,

g(n)=qrms _(j) (where jM≦n≦(j+1)M),

[0114] where g₁ (n) is g(n) of the current frame FR₁ and g₀ (n) is g(n)of the immediately preceding frame FR₀. In FIG. 6, each frame is dividedinto eight sub-frames SB (M=8)

[0115] Since analysis window w ((N/2)−1˜n) is placed on the data of thelatter half of the immediately preceding frame FR₀ for MDCT after thesubtraction using go (n+(N/2)) for the purpose of gain control at theside of the encoder, the signal obtained by placing analysis windoww((N/2)−1˜n), which is the sum P(n) of the principal component and thealiasing component, after inverse MDCT at the side of the decoder isexpressed by formula (2) below. $\begin{matrix}\begin{matrix}{{P(n)} = \quad {{{w\left( {\frac{N}{2} - 1 - n} \right)}{w\left( {\frac{N}{2} - 1 - n} \right)}\frac{1}{g_{0}\left( {n + \frac{N}{2}} \right)}{x(n)}} +}} \\{\quad {{w(n)}{w\left( {\frac{N}{2} - 1 - n} \right)}\frac{1}{g_{0}\left( {N - 1 - n} \right)}{x\left( {\frac{N}{2} - 1 - n} \right)}}}\end{matrix} & (2)\end{matrix}$

[0116] Additionally, analysis window w(n) is placed on the data of theformer half of the current frame FR₁ for MDCT after the subtractionusing g₀ (n) for the purpose of gain control at the side of the encoder,the signal obtained by placing analysis window w(n), which is the sumeQ(n) of the principal component and the aliasing component, afterinverse MDCT at the side of the decoder is expressed by formula (3)below. $\begin{matrix}\begin{matrix}{{Q(n)} = \quad {{{w(n)}{w(n)}\frac{1}{g_{1}(n)}{x(n)}} -}} \\{\quad {\left( {\frac{N}{2} - 1 - n} \right){w(n)}\frac{1}{g_{1}\left( {\frac{N}{2} - 1 - n} \right)}{x\left( {\frac{N}{2} - 1 - n} \right)}}}\end{matrix} & (3)\end{matrix}$

[0117] Therefore, x(n) to be reproduced can be obtained by formula (4)below. $\begin{matrix}{{x(n)} = \frac{\frac{P(n)}{g_{1}\left( {\frac{N}{2} - 1 - n} \right)} + \frac{Q(n)}{g_{0}\left( {N - 1 - n} \right)}}{\begin{matrix}{{{w\left( {\frac{N}{2} - 1 - n} \right)}{w\left( {\frac{N}{2} - 1 - n} \right)}\frac{1}{g_{0}\left( {n + \frac{N}{2}} \right)}} +} \\{{w(n)}{w(n)}\frac{1}{{g_{0}\left( {N - 1 - n} \right)}{g_{1}(n)}}}\end{matrix}}} & (4)\end{matrix}$

[0118] Thus, by placing windows in a manner as described below andcarrying out gain control operations using the rms of each sub-block(sub-frame) as envelope, the quantization noise such as pre-echo that isharsh to the human ear can be reduced relative to a sound that changesquickly with time, a tune having an acute attack or sound that quicklyattenuates from peak to peak.

[0119] Then, the MDCT coefficient data obtained by the MDCT operation ofthe MDCT circuit 27 of the orthogonal transform circuit section 25 aresent to the frame gain normalization circuit 43 and the frame gaincalculation/quantization circuit 47 of the coefficient quantizationsection 40. The coefficient quantization section 40 of this embodimentfirstly calculate the frame gain (block gain) of the entire coefficientsof a frame, which is an MDCT transform block, and normalizes the gain.Then, it divides it into critical bands, or sub-bands of which a bandwith a higher pitch level has a greater width as in the case of thehuman hearing sense, computationally determines the Bark scale factorfor each band and carries out a normalizing operation once again byusing the obtained scale factor. The value that can be used for the Barkscale factor may be the peak value of the coefficients within each bandor the square mean root (rms) of the coefficients. The Bark scalefactors of the bands are collectively vector-quantized.

[0120] More specifically, the frame gain calculation/quantizationcircuit 47 of the coefficient quantization section 40 computationallydetermines and quantizes the gain of each frame, which is an MDCTtransform block as described above and the obtained code book index(frame gain index) is taken out by way of terminal 55 and sent to thedecoder, while the frame gain of the quantized value is sent to theframe gain normalization circuit 43, which normalizes the value bydividing the input by the former. The output normalized by the framegain is then sent to the Bark scale factor calculation/quantizationcircuit 42 and the Bark scale factor normalization circuit 44.

[0121] The Bark scale factor calculation/quantization circuit 42computationally determines and quantizes the Bark scale factor of eachcritical band, which scale factor is then taken out by way of terminal54 and sent to the decoder. At the same time, the quantized Bark scalefactor is sent to the bit allocation calculation circuit 41 and the Barkscale factor normalization circuit 44. The Bark scale factornormalization circuit 44 normalizes the coefficients of each criticalband and the coefficients normalized by means of the Bark scale factorare sent to the coefficient quantization circuit 45.

[0122] In the coefficient quantization circuit 45, a given number ofbits are allocated to each coefficient according to the bit allocationinformation sent from the bit allocation calculation circuit 41. At thistime, the overall number of the allocated bits is switched according tothe gain control SW information sent from the above described gaincontrol on/off judgment circuit 18. In the case of vector-quantization,this arrangement can be realized by preparing two different code books,one for the on state of gain control and the other for the off state ofgain control, and selectively using either of them according to the gaincontrol switch information.

[0123] Now, the operation of bit allocation of the bit allocationcalculation circuit 41 will be described. Firstly, the weight to be suedfor quantizing each MDCT coefficient is computationally determined bymeans of the LPC coefficients, the pitch parameters or the Bark scalefactors obtained in a manner as described above. Then, the number ofbits to be allocated to each and every MDCT coefficient of the entirebands is determined and the MDCT coefficient is quantized. Thus, theweight can be regarded as noise-shaping factor and made to show desirednoise-shaping characteristics by modifying each of the parameters. As anexample, weights W(ω) are computationally determined by using only LPCcoefficients, pitch parameters and Bark scale factors as expressed byformulas below.

W(ω)=H(ω)P(ω)S(ω)

[0124] where H(ω) and P(ω) are frequency responses of transfer functionsH(z) and P(z),${H(z)} = \frac{1 + {\sum\limits_{i = 1}^{20}{\gamma^{i}\alpha_{i}z^{- i}}}}{1 + {\sum\limits_{i = 1}^{20}{\lambda^{i}\alpha_{i}z^{- i}}}}$

[0125] (weight obtained by using LPC coefficients)

[0126] γ=0.9, γ=0.8${P(z)} = \frac{1}{1 + {\sum\limits_{i = {- 1}}^{1}{\mu \quad g_{i}z^{{- k} + i}}}}$

[0127] (weight obtained by using pitch parameters)

[0128] μ=0.9

S(ω)=rms ₁(ω⊂bark_(i))  (5)

[0129] (weight obtained by using Bark scale factors)

[0130] Thus, the weights to be used quantization are determined by usingonly LPC coefficients, pitch coefficients or Bark scale factors so thatit is sufficient for the encoder to transit the parameters of the abovethree types to the decoder to make the latter reproduce the bitallocation of the encoder without transmitting any other bit allocationinformation so that the rate of transmitting side information can bereduced.

[0131] Now the quantizing operation of the coefficient quantizationcircuit 45 will be described by way of an example illustrated in FIGS.1B, 7A through 7C and 8.

[0132]FIG. 1B is a schematic block diagram of an exemplary coefficientquantization circuit 45 shown in FIG. 2. Normalized coefficient data(e.g., MDCT coefficients) y ae fed from the Bark scale factornormalization circuit 44 of FIG. 2 to input terminal 1. Weightcalculation circuit 2 is substantially equal to the bit allocationcalculation circuit 41 of FIG. 2. To be more accurate, it is realized bytaking out the portion adapted to calculate the weights to be used forallocating quantization bits out of the latter. The weight calculationcircuit 2 computationally determines the weights to be used for bitallocation on the basis of LPC coefficients, pitch parameters and Barkscale factors. Note that the coefficient of a frame is expressed byvector y and the weight of the frame is expressed by vector w.

[0133]FIGS. 7A through 7C are schematic illustrations of a sortingoperation based on the weights of coefficients within a band obtained bydividing coefficient data. FIG. 7A shows the weight vector w_(k) of thek-th band and FIG. 7B shows the coefficient vector y_(k) of the k-thband. In FIGS. 7A through 7C, the k-th band contains a total of eightelements and the eight weights that are the elements of the weightvector w_(k) are expressed respectively by w₁, w₂, . . . , w₈, whereasthe eight coefficients that ae the elements of the coefficient vectory_(k) are expressed respectively by y₁, y₂, . . . , y₈. In the exampleof FIGS. 7A and 7B, the weight W₃ corresponding to the coefficient y₃has the greatest value of all and followed by the remaining weights thatcan be arranged in the descending order of w₂, w₆, . . . , w₄. Then, thecoefficients y₁, y₂, . . . , y₈ are rearranged (sorted) to thecorresponding order of y₃, y₂, y₆, . . . , y₄. FIG. 7C shows thecollective coefficient vector of y′_(k).

[0134] Then, the coefficient vectors y′₀, y′₁, . . . , y′_(L−1) of therespective bands that are sorted in the descending order of thecorresponding weights are sent to the respective vector quantizers 5 ₀,5 ₁, . . . , 5 _(L−1) for vector-quantization. Preferably, the number ofbits allocated to each of the bands is preselected so that the number ofquantization bits allocated to each band may not fluctuate if the energyof the band changes.

[0135] As for the operation of vector-quantization, if the number ofelements of each band is large, they may be divided into a number ofsub-vectors and the operation of vector-quantization may be carried outfor each sub-vector. In other words, after sorting the coefficientvectors of the k-th band, the coefficient vector y′_(k) is divided intoa number of sub-vectors as shown in FIG. 8, the number being equal tothe predetermined number of elements. If the number is equal to three,the coefficient vector y′_(k) will be divided into three sub-vectorsy′_(k1), y′_(k2), y′_(k3), each of which is then vector-quantized toobtain code book indexes c_(k1), c_(k2), c_(k3). The indexes c_(k1),c_(k2), c_(k3) of the k-th band is collectively expressed by vectorc_(k). The operation of quantizing the sub-vectors can be carried out inthe descending order of the weights by allocating more quantization bitsto a vector located closer to the leading vector. In FIG. 8, forexample, the sub-vectors y′_(k1), y′_(k2), y′_(k3) can be arranged inthe descending order without changing the current order by allocating 8bits to the sub-vector y′_(k1), 6 bits to the sub-vector y′_(k2) and 4bits to the sub-vector y′_(k3). In other words, bits are allocated inthe descending order of the weights.

[0136] Then, the vectors c₀, c₁, . . . , C_(L−1) of the coefficientindexes of each band obtained from the respective vector quantizer 5 ₀,5 ₁, . . . , 5 _(L−1) are collectively taken out by way of terminal 6 asvector c of the coefficient indexes of all the bands. Note that theterminal 6 corresponds to the terminal 51 of FIG. 2.

[0137] In the example of FIGS. 1B, 7A through 7C and 8, the orthogonallytransformed coefficients on the frequency base (e.g., MDCT coefficients)are sorted by means of above described weights and rearranged in thedescending order of the numbers of allocated bits (so that a coefficientlocated close to the leading coefficient is allocated with a largernumber of bits). However, alternatively, only the indexes indicating thepositions or the order of the coefficients on the frequency baseobtained through orthogonal transform may be sorted in the descendingorder of said weights and the accuracy quantization of each coefficient(the number of bits allocated to it) may be determined as a function ofthe corresponding indexes. While vector quantization is used forquantizing the coefficients in the above described example, the presentinvention can alternatively be applied to an operation of scalarquantization or that of quantization using both scalars and vectors.

[0138] Now, an embodiment of audio signal decoder that corresponds tothe audio signal encoder of FIG. 2 will be described by referring toFIG. 9.

[0139] In FIG. 9, input terminals 60 through 67 are fed with data fromthe corresponding respective output terminals of FIG. 2. Morespecifically, the input terminal 60 of FIG. 9 is fed with indexes oforthogonal transform coefficients (e.g., MDCT coefficients) from theoutput terminal 51. Similarly, the input terminal 61 is fed with LSPindexes from the output terminal 31 of FIG. 2. The input terminals 62through 65 are fed respectively with data, or pitch lag indexes, pitchgain indexes, Bark scale factors and frame gain indexes from thecorresponding respective output terminals 52 through 55 of FIG. 2.Likewise, the input terminals 66, 67 are fed respectively with envelopeindexes and gain control SW information from the correspondingrespective output terminals 21, 22 of FIG. 2.

[0140] The coefficient indexes sent from the input terminal 60 areinversely quantized by coefficient inverse quantization circuit 71 andsent to inverse orthogonal transform circuit 74 for IMDCT (inverse MDCT)by way of multiplier 73.

[0141] The LSP indexes sent from the input terminal 61 are sent toinverse quantizer5 81 of LPC parameter reproduction section 80 andinversely quantized to LSP data by the section 80 and the output of thesection 80 is sent to LSP→α transform circuit 82 and LSP interpolationcircuit 83. The α parameters (LPC coefficients) from the LSP→α transformcircuit 82 are sent to bit allocation circuit 72. The LSP data from theLSP interpolation circuit 83 are transformed into α parameters (LPCcoefficients) by LSP→α transform circuit 84 and sent to LPC synthesiscircuit 77.

[0142] The bit allocation circuit 72 is supplied with pitch lags fromthe input terminal 62, pitch gains from the input terminal 63 coming byway of inverse quantizer 91 and Bark scale factors from the inputterminal 64 coming by way of inverse quantizer 92 in addition to saidLPC coefficients from the LSP→α transform circuit 82. Then, the decodercan reproduce the bit allocation of the encoder only on the basis of theparameters. The bit allocation information from the bit allocationcircuit 72 is sent to coefficient inverse quantizer 71, which uses theinformation for determining the number of bits allocated to eachcoefficient for quantization.

[0143] The frame gain indexes from the input terminal 65 are sent toframe gain inverse quantizer 86 and inversely quantized. The obtainedframe gain is then sent to multiplier 73.

[0144] The envelope index from the input terminal 66 is sent to envelopeinverse quantizer 88 by way of switch 87 and inversely quantized. Theobtained envelope data are then sent to overlapped addition circuit 75.The gain control SW information from the input terminal 67 is sent tothe coefficient inverse quantizer 71 and the overlapped addition circuit75 and also used as control signal for the switch 87. Said coefficientinverse quantizer 71 switches the total number of bits to be allocateddepending on the on/off state of the above described gain control. Inthe case of inverse quantization, two different code books may beprepared, one for the on state of gain control and the other for the offstate of gain control, and selectively used according to the gaincontrol switch information.

[0145] The overlapped addition circuit 75 causes the signal that isbrought back to the time base on a frame by frame basis and sent fromthe inverse orthogonal transform circuit 7 typically for IMDCT to beoverlapped by ½ frame for each frame and adds the frames. When the gaincontrol is on, it performs the operation of overlapped addition whileprocessing the gain control (gain expansion or gain restoration asdescribed earlier) by means of the envelope data from the envelopeinverse quantizer 88.

[0146] The time base signal from the overlapped addition circuit 75 issent to pitch synthesis circuit 76, which restores the pitch component.This operation is a reverse of the operation of the pitch inverse filter13 of FIG. 2 and the pitch lag from the terminal 62 and the pitch gainfrom the inverse quantizer 91 are used for this operation.

[0147] The output of the pitch synthesis circuit 76 is sent to the LPCsynthesis circuit 77, which carries out an operation of LPC synthesisthat is a reverse of the operation of the LPC inverse filter 12 of FIG.2. The outcome of the operation is taken out from output terminal 78.

[0148] If the coefficient quantization circuit 45 of the coefficientquantization section 40 of the encoder has a configuration adapted tovector-quantize the coefficients that are sorted for each band accordingto the allocated weights as shown in FIG. 7 (?), the coefficient inversequantization circuit 71 may have the configuration shown in FIG. 10

[0149] Referring to FIG. 10, input terminal 60 corresponds to the inputterminal of FIG. 9 and is fed with coefficient indexes (code bookindexes obtained by quantizing orthogonal transform coefficients such asMDCT coefficients), whereas weight calculation circuit 79 is fed with αparameters (LPC coefficients) from the LSP→α transform circuit 82 ofFIG. 9, pitch lags from input terminal 62, pitch gains from the inversequantizer 91 and Bark scale factors from the inverse quantizer 92. Theweight calculation circuit 79 computationally determines weights W(ω) byusing only LPC coefficients, pitch parameters (pitch lags and pitchgains) and Bark scale factors in addition to the equation (5) above. Theinput terminal 92 is fed with numerical values of 0˜N−1 (which areexpressed by vector I) when there are indexes indicating the positionsor the order of arrangement of the coefficients on the frequency baseand hence there are a total of N coefficient data over the entire bands.Note that the N weights sent from the weight calculation circuit 79 forthe N coefficients are expressed by vector w.

[0150] The weight w from the weight calculation circuit 79 and the indexI from the input terminal 92 are sent to band dividing circuit 97, whichdivides each of them into L bands as in the case of the encoder. Ifthree bands of a low band, a middle band and a high band (L=3) are usedin the encoder, the band is divided into three bands also in thedecoder. Then, the indexes and the weights of the three bands arerespectively sent to sorting circuits 95 ₀, 95 ₁, . . . , 95 _(L−1). Forexample, index I_(k) and weight w_(k) of the k-th band. In the sortingcircuit 95 _(k), the index I_(k) in the k-th band are rearranged(sorted) according to the order of arrangement of the weights w_(k) ofthe coefficients and the sorted index I′_(k) are output. The sortedindex I₀, I₁, . . . , I_(L−1) sorted for each band by the respectivesorting circuits 95 ₀, 95 ₁, . . . , 95 _(L−1) are then sent tocoefficient reorganization circuit 97.

[0151] The indexes of the orthogonal coefficients from the inputterminal 60 are obtained during the quantizing operation of the encoderin such a way that the original band is divided into L bands and thecoefficients are sorted in the descending order of the weights in eachband and vector-quantized for each of the sub-vectors obtained accordingto a predetermined rule in the band. More specifically, the sets ofcoefficient indexes of each of a total of L bands are expressedrespectively by vectors c₀, c₁, . . . , c_(L−1), which are then sent torespective inverse quantizers 95 ₀, 95 ₁, . . . , 95 _(L−1). Thecoefficient data obtained by the inverse quantizers 95 ₀, 95 ₁, . . . ,95 _(L−1) as a result of inverse quantization correspond to those thatare sorted in the descending order of the weights in each band, or thecoefficient vectors y′₀, y′₁, . . . , y′_(L−1) from the sorting circuits4 ₀, 4 ₁, . . . , 4 _(L−1) as shown in FIG. 1B so that the order orarrangement is different from that of arrangement on the frequency base.Thus, the coefficient reorganization circuit 97 is adapted to sort theindexes I in advance in the descending order of the weights and restoresthe original order on the frequency base by making the sorted indexescorrespond to the respective coefficient data obtained by the aboveinverse quantization. In short, the coefficient reorganization circuit97 retrieves the coefficient data y showing the original order ofarrangement on the frequency base by making the sorted indexes from thesorting circuits 95 ₀, 95 ₁, . . . , 95 _(L−1) correspond to therespective coefficient data from the inverse quantizers 96 ₀, 96 ₁, . .. , 96 _(L−1) that are sorted in the descending order of the weights ineach band and rearranging (inversely sorting) the coefficient dataaccording to the sorted indexes and then it takes out the coefficientdata y from output terminal 98. The coefficient data from the outputterminal 98 are then sent to the multiplier 73 in FIG. 9.

[0152]FIG. 11 is a schematic block diagram of an embodiment of decodercorresponding to the encoder of FIG. 1C.

[0153] Referring to FIG. 12, input terminal 60 and input terminal 66 arerespectively fed with coefficient indexes and envelope indexes, whichare described above. The coefficient indexes of the input terminal 60are then inversely quantized by inverse quantization circuit 71 andprocessed for inverse MDCT (inverse orthogonal transform) by IMDCTcircuit before sent to overlapped addition circuit 75. The envelopeindexes of the input terminal 66 are then inversely quantized by inversequantizer 88 and the envelope information is sent to the overlappedaddition circuit 75. The overlapped addition circuit 75 carries out anoperation that is reverse to the above described gain smoothingoperation (of dividing the input signal with the envelope information bymeans of the divider 14) and also an operation of overlapped addition inorder to output a continuous time base signal from terminal 89. Thesignal from the terminal 89 is sent to the pitch synthesis circuit 76 ofFIG. 9.

[0154] With the above described processing, the signal is subjected to anoise shaping operation along the time base so that any quantizationnoise that is harsh to the human ear can be reduced without switch inthe transform window size.

[0155] As an example where the present invention is applied, FIG. 12shows a reproduced signal waveform that can be obtained by encoding asound of a castanet without gain control, whereas FIG. 13 shows areproduced signal waveform that can be obtained by encoding a sound of acastanet with gain control. As clearly seen from FIGS. 12 and 13, thenoise prior to the attack of a tune (so-called pre-echo) can beremarkably reduced by applying gain control according to the invention.

[0156]FIG. 14 shows the waveform of a time base signal in an initialstage of the speech burst of part of a sound signal, whereas FIG. 15shows the frequency spectrum in an initial stage of the speech burst ofpart of a sound signal. In each of FIGS. 14 and 15, the curve a showsthe use of gain control, whereas curve b (broken line) shows the non-useof gain control. By comparing the curves a and b, the curve a with theuse of gain control clearly shows the pitch structure and hence a goodreproduction performance as particularly clearly revealed in FIG. 15.

[0157] The present invention is by no means limited to the aboveembodiment. For example, the input time base signal may be a voicesignal in the telephone frequency band or a video signal and may not bean audio signal, which may be a voice signal or a music tone signal. Theconfiguration of the normalization circuit section 11, the LPC analysisand the pitch analysis are not limited to the above description and anyof various alternative techniques such as extracting and removing thecharacteristic traits or the correlation of the time base input waveformby means of linear prediction or non-linear prediction may be used forthe purpose of the invention. The quantizers may be scalar quantizers orscalar quantizers and vector quantizers may be combinedly used for thequantizers. They should not necessarily be vector quantizers.

What is claimed is:
 1. A signal coding apparatus comprising: anormalization means for removing the correlation of the signal waveformon the basis of the parameters obtained by carrying out linearprediction coding analysis and pitch analysis on the4 input signal onthe time base and taking out the residue; an orthogonal transform meansfor carrying out an orthogonal transform operation on the output of thenormalization means; and a quantization means for quantizing the outputof the orthogonal transform means.
 2. A signal coding apparatusaccording to claim 1, wherein said orthogonal transform means transformsthe time base signal input by modified discrete cosine transform (MDCT)into coefficient data on the frequency base.
 3. A signal codingapparatus according to claim 2, wherein said normalization meansincludes an LPC inverse filter for outputing the LPC prediction residueof said input signal on the basis of the LPC coefficients obtained byLPC analysis conducted on said input signal and a pitch inverse filterfor removing the correlation of the pitch of the LPC predeterminedresidue on the basis of the pitch parameters obtained by pitch analysisconducted on said LPC prediction residue.
 4. A signal coding apparatusaccording to claim 2, wherein said quantization means quantizesaccording to the number of allocated bits as determined on the basis ofthe LPC analysis and the pitch analysis.
 5. A signal coding method forencoding an input signal on the time base through orthogonal transform,said method comprising: a step of removing the correlation of signalwaveform on the basis of the parameters obtained by means of linearpredictive coding (LPC) analysis and pitch analysis of the input signalon the time base prior to the orthogonal transform.
 6. A signal codingmethod according to claim 5, wherein modified discrete cosine transform(MDCT) is used for the orthogonal transform.
 7. A signal codingapparatus comprising: an analysis means for analysing the input signalon the time base and extracting the characteristic traits of the signalwaveform; a normalization means for removing the correlation of saidinput signal on the basis of the analysis of the analysis means andtaking out the residue; a quantization means for quantizing the outputof the orthogonal transform means; and a bit allocation calculationmeans for determining the bit allocation for the quantization of saidquantization means on the basis of the analysis of said analysis means.8. A signal coding apparatus according to claim 7, wherein saidorthogonal transform means transforms the time base signal input bymodified discrete cosine transform (MDCT) into coefficient data on thefrequency base.
 9. A signal coding apparatus according to claim 8,wherein said analysis means includes an LPC analysis means for carryingout a linear predetermined coding (LPC) analysis on said input signaland outputting LPC coefficients and a pitch analysis means for carryingout a pitch analysis on the LPC prediction residue and outputting pitchparameters; and said normalization means includes an LPC inverse filterfor outputing the LPC prediction residue of said input signal on thebasis of the LPC coefficients from said LPC analysis means and a pitchinverse filter for removing the correlation of the pitch of the LPCprediction residue on the basis of the pitch parameters from said pitchanalysis means.
 10. A signal coding apparatus according to claim 9,wherein said bit allocation calculation means determines the bitallocation for the quantization of the coefficient output from saidorthogonal transform means on the basis of said LPC coefficients fromsaid LPC analysis means, said pitch parameters from said pitch analysismeans and the Bark scale factors obtained for each critical band of thecoefficient output of said orthogonal transform means.
 11. A signalcoding method for encoding an input signal on the time base throughorthogonal transform, said method comprising: a step of determining thebit allocation for the quantization of the coefficients obtained by saidorthogonal transmitter on the basis of linear prediction coding (LPC)analysis and pitch analysis.
 12. A signal coding method according toclaim 11, wherein modified discrete cosine transform (MDCT) is used forthe orthogonal transform.
 13. A signal coding method according to claim11, wherein Bark scale factors obtained for each critical band of thecoefficients obtained by said orthogonal transform are also used fordetermining the bit allocation.
 14. A signal coding apparatus forencoding an input signal on the time base through orthogonal transmitterusing an orthogonal transmitter means; said apparatus comprising: aweight calculation means for calculating weights in response to saidinput signal; and a quantization means for assigning an order to thecoefficient data from said orthogonal transform means in the descendingorder of the weights from said weight calculation means and quantizingthe coefficient data of a higher order with a higher degree ofprecision.
 15. A signal coding apparatus according to claim 14, whereinsaid quantization is conducted in such a way that more bits areallocated to the coefficient data of a higher order.
 16. A signal codingapparatus according to claim 14, wherein the coefficient data from saidorthogonal transform means are divided into a plurality of bands on thefrequency base and assigning an order to the coefficient data of eachband in the descending order of the weights independently from theremaining bands.
 17. A signal coding apparatus according to claim 14,wherein the coefficient data are divided into groups from thecoefficient data of higher orders to form coefficient vectors and thecoefficient vectors are subsequently vector-quantized.
 18. A signalcoding apparatus according to claim 14, wherein said weight calculationmeans calculates said weights on the basis of the parametersrepresenting the statistic characteristics of the input signal includinglinear or non-linear analysis of the input signal.
 19. A signal codingapparatus according to claim 14, wherein said orthogonal transform meanstransforms the time base signal input by modified discrete cosinetransform (MDCT) into coefficient data on the frequency base.
 20. Asignal coding apparatus according to claim 14, wherein a normalizationmeans for removing the correlation of the signal waveform of the inputsignal and taking out the residue is arranged on the input side of saidorthogonal transform means; said normalization means including an LPCinverse filter for outputing the LPC prediction residue of said inputsignal on the basis of the LPC coefficients obtained by LPC analysisconducted on said input signal and a pitch inverse filter for removingthe correlation of the pitch of the LPC predetermined residue on thebasis of the pitch parameters obtained by pitch analysis conducted onsaid LPC prediction residue; said weight calculation means calculatingsaid weights on the basis of said LPC coefficients and said pitchparameters.
 21. A signal coding method for encoding an input signal onthe time base through orthogonal transform, said method comprising: aweight calculation step of calculating weights in response to the inputsignal; and a quantization step of assigning an order to the coefficientdata from said orthogonal transform means in the descending order of theweights from said weight calculation means and quantizing thecoefficient data of a higher order with a higher degree of precision.22. A signal coding method according to claim 21, wherein saidquantization is conducted in such a way that more bits are allocated tothe coefficient data of a higher order.
 23. A signal coding methodaccording to claim 21, wherein the coefficient data obtained by saidorthogonal transform are divided into a plurality of bands on thefrequency base and an order is assigned to the coefficient data of eachband in the descending order of the weights independently from theremaining bands.
 24. A signal coding method according to claim 21,wherein the coefficient data are divided into groups from thecoefficient data of higher orders to form coefficient vectors and thecoefficient vectors are subsequently vector-quantized.
 25. A signaldecoding apparatus for decoding coded data obtained by performingorthogonal transform on an input signal on the time base and quantizingthe coefficient data obtained by the orthogonal transform with differentlevels of precision of quantization selected according to the weightsdetermined on the basis of the input signal, said decoding apparatuscomprising; a weight calculation means for calculating the weights usedfor the coding on the basis of the parameters input for the weightcalculation; and an inverse quantization means for inversely quantizingthe data obtained by quantizing said coefficient data with the differentlevels of precision of quantization selected according to the weightsdetermined by said weight calculation means.
 26. A signal decodingapparatus according to claim 25, wherein said weights are calculated onthe basis of the parameters representing the statistic characteristicsof the input signal including linear or non-linear analysis of the inputsignal. said parameters being transmitted and supplied; said weightcalculation means calculating said weights on the basis of the suppliedparameters.
 27. A signal decoding method for decoding coded dataobtained by performing orthogonal transform on an input signal on thetime base and quantizing the coefficient data obtained by the orthogonaltransform with different levels of precision of quantization selectedaccording to the weights determined on the basis of the input signal,said decoding method comprising; a weight calculation step ofcalculating the weights used for the coding on the basis of theparameters input for the weight calculation; and an inverse quantizationstep of inversely quantizing the data obtained by quantizing saidcoefficient data with the different levels of precision of quantizationselected according to the weights determined by said weight calculationmeans.
 28. A signal coding apparatus for coding an input signal on thetime base frame by frame, said frames being used as coding units,through orthogonal transform by means of an orthogonal transmittermeans, said coding apparatus comprising: an envelope extraction meansfor extracting the envelope in each frame of said input signal; and again smoothing means for gain-smoothing on said input signal on thebasis of the envelope extracted by said envelope extraction means andsupplying it to said orthogonal transform means.
 29. A signal codingapparatus according to claim 28, wherein the envelope information fromsaid envelope extraction means is quantized, output and subjected to again smoothing operation by means of said quantized envelope.
 30. Asignal coding apparatus according to claim 28, wherein said envelopeextraction means calculates as said envelope the square means root (rms)of each sub-frame produced by dividing said frame into a plurality ofsub-frames.
 31. A signal coding apparatus according to claim 30, whereinsaid rms of each sub-frame is quantized and output and saidgain-smoothing operation is performed on the basis of the rms of eachsub-frame.
 32. A signal coding apparatus according to claim 28, whereinsaid orthogonal transform means transforms the time base signal input bymodified discrete cosine transform (MDCT) into coefficient data on thefrequency data.
 33. A signal coding apparatus according to claim 28,wherein a normalization means is connected to the upstream of saidorthogonal transform means and said normalization means includes an LPCinverse filter for outputing the LPC prediction residue of said inputsignal on the basis of the LPC coefficients obtained by LPC analysisconducted on said input signal and a pitch inverse filter for removingthe correlation of the pitch of the LPC predetermined residue on thebasis of the pitch parameters obtained by pitch analysis conducted onsaid LPC prediction residue.
 34. A signal coding apparatus according toclaim 33, wherein said quantization means quantizes according to thenumber of allocated bits as determined on the basis of the LPC analysisand the pitch analysis.
 35. A signal coding method for coding an inputsignal on the time base frame by frame, said frames being used as codingunits, through orthogonal transform by means of an orthogonaltransmitter means, said coding method comprising: an envelope extractionstep of extracting the envelope in each frame of said input signal; anda gain smoothing step of gain-smoothing on said input signal on thebasis of the envelope extracted by said envelope extraction means andsupplying it to said orthogonal transform means.
 36. A signal codingmethod according to claim 35, wherein the square means root (rms) ofeach sub-frame produced by dividing said frame into a plurality ofsub-frames is calculated as said envelope in said envelope extract step.37. A signal coding method according to claim 35, wherein the time basesignal input is transformed by modified discrete cosine transform (MDCT)into coefficient data on the frequency data in said orthogonal transformstep.
 38. A signal decoding apparatus for extracting an envelope for aninput signal on the time base frame by frame, said frames being used ascoding units, gain-smoothing said input signal on the basis of theextracted envelope, supplying coded data obtained by performingorthogonal transform and coding on the gain-smoothed signal and decodingthe coded data, said decoding apparatus comprising: an inverseorthogonal transform means for inversely transforming said coded data;and an overlapped addition means for performing an overlapped additionon the signal subjected to the inverse orthogonal transform, whileperforming an inverse gain-smoothing operation on the signal, tocontinuously output a time base signal.
 39. A signal decoding method forextracting an envelope for an input signal on the time base frame byframe, said frames being used as coding units, gain-smoothing said inputsignal on the basis of the extracted envelope, supplying coded dataobtained by performing orthogonal transform and coding on thegain-smoothed signal and decoding the coded data, said decoding methodcomprising: an inverse orthogonal transform step of inverselytransforming said coded data; and an overlapped addition step ofperforming an overlapped addition on the signal subjected to the inverseorthogonal transform, while performing an inverse gain-smoothingoperation on the signal, to continuously output a time base signal.